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Linshizhi
ffmpeg.wasm-core
Commits
323d3752
Commit
323d3752
authored
Oct 11, 2015
by
Claudio Freire
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AAC encoder: cosmetics from last commit
Reindent
parent
01ecb717
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Showing
2 changed files
with
21 additions
and
21 deletions
+21
-21
aacpsy.c
libavcodec/aacpsy.c
+9
-9
psymodel.c
libavcodec/psymodel.c
+12
-12
No files found.
libavcodec/aacpsy.c
View file @
323d3752
...
@@ -708,16 +708,16 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
...
@@ -708,16 +708,16 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
pctx
->
pe
.
max
=
FFMAX
(
pe
,
pctx
->
pe
.
max
);
pctx
->
pe
.
max
=
FFMAX
(
pe
,
pctx
->
pe
.
max
);
pctx
->
pe
.
min
=
FFMIN
(
pe
,
pctx
->
pe
.
min
);
pctx
->
pe
.
min
=
FFMIN
(
pe
,
pctx
->
pe
.
min
);
}
else
{
}
else
{
desired_bits
=
calc_bit_demand
(
pctx
,
pe
,
ctx
->
bitres
.
bits
,
ctx
->
bitres
.
size
,
wi
->
num_windows
==
8
);
desired_bits
=
calc_bit_demand
(
pctx
,
pe
,
ctx
->
bitres
.
bits
,
ctx
->
bitres
.
size
,
wi
->
num_windows
==
8
);
desired_pe
=
PSY_3GPP_BITS_TO_PE
(
desired_bits
);
desired_pe
=
PSY_3GPP_BITS_TO_PE
(
desired_bits
);
/* NOTE: PE correction is kept simple. During initial testing it had very
/* NOTE: PE correction is kept simple. During initial testing it had very
* little effect on the final bitrate. Probably a good idea to come
* little effect on the final bitrate. Probably a good idea to come
* back and do more testing later.
* back and do more testing later.
*/
*/
if
(
ctx
->
bitres
.
bits
>
0
)
if
(
ctx
->
bitres
.
bits
>
0
)
desired_pe
*=
av_clipf
(
pctx
->
pe
.
previous
/
PSY_3GPP_BITS_TO_PE
(
ctx
->
bitres
.
bits
),
desired_pe
*=
av_clipf
(
pctx
->
pe
.
previous
/
PSY_3GPP_BITS_TO_PE
(
ctx
->
bitres
.
bits
),
0
.
85
f
,
1
.
15
f
);
0
.
85
f
,
1
.
15
f
);
}
}
pctx
->
pe
.
previous
=
PSY_3GPP_BITS_TO_PE
(
desired_bits
);
pctx
->
pe
.
previous
=
PSY_3GPP_BITS_TO_PE
(
desired_bits
);
ctx
->
bitres
.
alloc
=
desired_bits
;
ctx
->
bitres
.
alloc
=
desired_bits
;
...
...
libavcodec/psymodel.c
View file @
323d3752
...
@@ -111,18 +111,18 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
...
@@ -111,18 +111,18 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
/* AAC has its own LP method */
/* AAC has its own LP method */
if
(
avctx
->
codec_id
!=
AV_CODEC_ID_AAC
)
{
if
(
avctx
->
codec_id
!=
AV_CODEC_ID_AAC
)
{
if
(
avctx
->
cutoff
>
0
)
if
(
avctx
->
cutoff
>
0
)
cutoff_coeff
=
2
.
0
*
avctx
->
cutoff
/
avctx
->
sample_rate
;
cutoff_coeff
=
2
.
0
*
avctx
->
cutoff
/
avctx
->
sample_rate
;
if
(
cutoff_coeff
&&
cutoff_coeff
<
0
.
98
)
if
(
cutoff_coeff
&&
cutoff_coeff
<
0
.
98
)
ctx
->
fcoeffs
=
ff_iir_filter_init_coeffs
(
avctx
,
FF_FILTER_TYPE_BUTTERWORTH
,
ctx
->
fcoeffs
=
ff_iir_filter_init_coeffs
(
avctx
,
FF_FILTER_TYPE_BUTTERWORTH
,
FF_FILTER_MODE_LOWPASS
,
FILT_ORDER
,
FF_FILTER_MODE_LOWPASS
,
FILT_ORDER
,
cutoff_coeff
,
0
.
0
,
0
.
0
);
cutoff_coeff
,
0
.
0
,
0
.
0
);
if
(
ctx
->
fcoeffs
)
{
if
(
ctx
->
fcoeffs
)
{
ctx
->
fstate
=
av_mallocz
(
sizeof
(
ctx
->
fstate
[
0
])
*
avctx
->
channels
);
ctx
->
fstate
=
av_mallocz
(
sizeof
(
ctx
->
fstate
[
0
])
*
avctx
->
channels
);
for
(
i
=
0
;
i
<
avctx
->
channels
;
i
++
)
for
(
i
=
0
;
i
<
avctx
->
channels
;
i
++
)
ctx
->
fstate
[
i
]
=
ff_iir_filter_init_state
(
FILT_ORDER
);
ctx
->
fstate
[
i
]
=
ff_iir_filter_init_state
(
FILT_ORDER
);
}
}
}
}
ff_iir_filter_init
(
&
ctx
->
fiir
);
ff_iir_filter_init
(
&
ctx
->
fiir
);
...
...
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